Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: Informational Ericsson
Expires: January 16, 2014 C. Perkins
University of Glasgow
H. Alvestrand
Google
July 15, 2013
Guidelines for using the Multiplexing Features of RTP to SupportMultiple Media Streamsdraft-ietf-avtcore-multiplex-guidelines-01
Abstract
The Real-time Transport Protocol (RTP) is a flexible protocol that
can be used in a wide range of applications, networks, and system
topologies. That flexibility makes for wide applicability, but can
complicate the application design process. One particular design
question that has received much attention is how to support multiple
media streams in RTP. This memo discusses the available options and
design trade-offs, and provides guidelines on how to use the
multiplexing features of RTP to support multiple media streams.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 16, 2014.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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RTP was from the beginning designed for multiple participants in a
communication session. This is not restricted to multicast, as some
believe, but also provides functionality over unicast, using either
multiple transport flows below RTP or a network node that re-
distributes the RTP packets. The re-distributing node can for
example be a transport translator (relay) that forwards the packets
unchanged, a translator performing media or protocol translation in
addition to forwarding, or an RTP mixer that creates new sources from
the received streams. In addition, multiple streams can occur when a
single endpoint have multiple media sources, like multiple cameras or
microphones that need to be sent simultaneously.
This document has been written due to increased interest in more
advanced usage of RTP, resulting in questions regarding the most
appropriate RTP usage. The limitations in some implementations, RTP/
RTCP extensions, and signalling has also been exposed. It is
expected that some limitations will be addressed by updates or new
extensions resolving the shortcomings. The authors also hope that
clarification on the usefulness of some functionalities in RTP will
result in more complete implementations in the future.
The document starts with some definitions and then goes into the
existing RTP functionalities around multiplexing. Both the desired
behaviour and the implications of a particular behaviour depend on
which topologies are used, which requires some consideration. This
is followed by a discussion of some choices in multiplexing behaviour
and their impacts. Some archetypes of RTP usage are discussed.
Finally, some recommendations and examples are provided.
2. Definitions2.1. Terminology
The following terms and abbreviations are used in this document:
Endpoint: A single entity sending or receiving RTP packets. It can
be decomposed into several functional blocks, but as long as it
behaves a single RTP stack entity it is classified as a single
endpoint.
Multiparty: A communication situation including multiple endpoints.
In this document it will be used to refer to situations where more
than two endpoints communicate.
Media Source: The source of a stream of data of one Media Type, It
can either be a single media capturing device such as a video
camera, a microphone, or a specific output of a media production
function, such as an audio mixer, or some video editing function.
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Sending data from a Media Source can cause multiple RTP sources to
send multiple Media Streams.
Media Stream: A sequence of RTP packets using a single SSRC that
together carries part or all of the content of a specific Media
Type from a specific sender source within a given RTP session.
RTP Source: The originator or source of a particular Media Stream.
Identified using an SSRC in a particular RTP session. An RTP
source is the source of a single media stream, and is associated
with a single endpoint and a single Media Source. An RTP Source
is just called a Source in RFC 3550.
RTP Sink: A recipient of a Media Stream. The Media Sink is
identified using one or more SSRCs. There can be more than one
RTP Sink for one RTP source.
CNAME: "Canonical name" - identifier associated with one or more RTP
sources from a single endpoint. Defined in the RTP specification
[RFC3550]. A CNAME identifies a synchronisation context. A CNAME
is associated with a single endpoint, although some RTP nodes will
use an endpoint's CNAME on that endpoints behalf. An endpoint can
use multiple CNAMEs. A CNAME is intended to be globally unique
and stable for the full duration of a communication session.
[RFC6222][I-D.ietf-avtcore-6222bis] gives updated guidelines for
choosing CNAMEs.
Media Type: Audio, video, text or data whose form and meaning are
defined by a specific real-time application.
Multiplexing: The operation of taking multiple entities as input,
aggregating them onto some common resource while keeping the
individual entities addressable such that they can later be fully
and unambiguously separated (de-multiplexed) again.
RTP Session: As defined by [RFC3550], the endpoints belonging to the
same RTP Session are those that share a single SSRC space. That
is, those endpoints can see an SSRC identifier transmitted by any
one of the other endpoints. An endpoint can receive an SSRC
either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP
Session scope is decided by the endpoints' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by endpoints and any interconnecting middle nodes.
RTP Session Group: One or more RTP sessions that are used together
to perform some function. Examples are multiple RTP sessions used
to carry different layers of a layered encoding. In an RTP
Session Group, CNAMEs are assumed to be valid across all RTP
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sessions, and designate synchronisation contexts that can cross
RTP sessions.
Source: Term that ought not be used alone. An RTP Source, as
identified by its SSRC, is the source of a single Media Stream; a
Media Source can be the source of mutiple Media Streams.
SSRC: A 32-bit unsigned integer used as identifier for a RTP Source.
CSRC: Contributing Source, A SSRC identifier used in a context, like
the RTP headers CSRC list, where it is clear that the Media Source
is not the source of the media stream, instead only a contributor
to the Media Stream.
Signalling: The process of configuring endpoints to participate in
one or more RTP sessions.
2.2. Subjects Out of Scope
This document is focused on issues that affect RTP. Thus, issues
that involve signalling protocols, such as whether SIP, Jingle or
some other protocol is in use for session configuration, the
particular syntaxes used to define RTP session properties, or the
constraints imposed by particular choices in the signalling
protocols, are mentioned only as examples in order to describe the
RTP issues more precisely.
This document assumes the applications will use RTCP. While there
are such applications that don't send RTCP, they do not conform to
the RTP specification, and thus can be regarded as reusing the RTP
packet format but not implementing the RTP protocol.
3. Reasons for Multiplexing and Grouping RTP Media Streams
The reasons why an endpoint might choose to send multiple media
streams are widespread. In the below discussion, please keep in mind
that the reasons for having multiple media streams vary and include
but are not limited to the following:
o Multiple Media Sources
o Multiple Media Streams might be needed to represent one Media
Source (for instance when using layered encodings)
o A Retransmission stream might repeat the content of another Media
Stream
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o An FEC stream might provide material that can be used to repair
another Media Stream
o Alternative Encodings, for instance different codecs for the same
audio stream
o Alternative formats, for instance multiple resolutions of the same
video stream
For each of these, it is necessary to decide if each additional media
stream gets its own SSRC multiplexed within a RTP Session, or if it
is necessary to use additional RTP sessions to group the media
streams. The choice between these made due to one reason might not
be the choice suitable for another reason. In the above list, the
different items have different levels of maturity in the discussion
on how to solve them. The clearest understanding is associated with
multiple media sources of the same media type. However, all warrant
discussion and clarification on how to deal with them. As the
discussion below will show, in reality we cannot choose a single one
of the two solutions. To utilise RTP well and as efficiently as
possible, both are needed. The real issue is finding the right
guidance on when to create RTP sessions and when additional SSRCs in
an RTP session is the right choice.
4. RTP Multiplexing Points
This section describes the multiplexing points present in the RTP
protocol that can be used to distinguish media streams and groups of
media streams. Figure 1 outlines the process of demultiplexing
incoming RTP streams:
|
| packets
+-- v
| +------------+
| | Socket |
| +------------+
| || ||
RTP | RTP/ || |+-----> SCTP ( ...and any other protocols)
Session | RTCP || +------> STUN (multiplexed using same port)
+-- ||
+-- ||
| (split by SSRC)
| || || ||
| || || ||
Media | +--+ +--+ +--+
Streams | |PB| |PB| |PB| Jitter buffer, process RTCP, FEC, etc.
| +--+ +--+ +--+
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+-- | | |
(pick rending context based on PT)
+-- | / |
| +---+ |
| / | |
Payload | +--+ +--+ +--+
Formats | |CR| |CR| |CR| Codecs and rendering
| +--+ +--+ +--+
+--
Figure 1: RTP Demultiplexing Process
4.1. RTP Session
An RTP Session is the highest semantic layer in the RTP protocol, and
represents an association between a group of communicating endpoints.
The set of participants that form an RTP session is defined as those
that share a single synchronisation source space [RFC3550]. That is,
if a group of participants are each aware of the synchronisation
source identifiers belonging to the other participants, then those
participants are in a single RTP session. A participant can become
aware of a synchronisation source identifier by receiving an RTP
packet containing it in the SSRC field or CSRC list, by receiving an
RTCP packet mentioning it in an SSRC field, or through signalling
(e.g., the SDP "a=ssrc:" attribute). Thus, the scope of an RTP
session is determined by the participants' network interconnection
topology, in combination with RTP and RTCP forwarding strategies
deployed by the endpoints and any middleboxes, and by the signalling.
RTP does not contain a session identifier. Rather, it relies on the
underlying transport layer to separate different sessions, and on the
signalling to identify sessions in a manner that is meaningful to the
application. The signalling layer might give sessions an explicit
identifier, or their identification might be implicit based on the
addresses and ports used. Accordingly, a single RTP Session can have
multiple associated identifiers, explicit and implicit, belonging to
different contexts. For example, when running RTP on top of UDP/IP,
an RTP endpoint can identify and delimit an RTP Session from other
RTP Sessions using the UDP source and destination IP addresses and
UDP port numbers. Another example is when using SDP grouping
framework [RFC5888] which uses an identifier per "m="-line; if there
is a one-to-one mapping between "m="-lines and RTP sessions, that
grouping framework identifier will identify an RTP Session.
RTP sessions are globally unique, but their identity can only be
determined by the communication context at an endpoint of the
session, or by a middlebox that is aware of the session context. The
relationship between RTP sessions depending on the underlying
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application, transport, and signalling protocol. The RTP protocol
makes no normative statements about the relationship between
different RTP sessions, however the applications that use more than
one RTP session will have some higher layer understanding of the
relationship between the sessions they create.
4.2. Synchronisation Source (SSRC)
A synchronisation source (SSRC) identifies an RTP source or an RTP
sink. Every endpoint will have at least one synchronisation source
identifier, even if it does not send media (endpoints that are only
RTP sinks still send RTCP, and use their synchronisation source
identifier in the RTCP packets they send). An endpoint can have
multiple synchronisation sources identifiers if it contains multiple
RTP sources (i.e., if it sends multiple media streams). Endpoints
that are both RTP sources and RTP sinks use the same synchronisation
sources in both roles. At any given time, a RTP source has one and
only one SSRC - although that can change over the lifetime of the RTP
source or sink.
The synchronisation Source identifier is a 32-bit unsigned integer.
It is present in every RTP and RTCP packet header, and in the payload
of some RTCP packet types. It can also be present in SDP signalling.
Unless pre-signalled using the SDP "a=ssrc:" attribute [RFC5576], the
synchronisation source identifier is chosen at random. It is not
dependent on the network address of the endpoint, and is intended to
be unique within an RTP session. Synchronisation source identifier
collisions can occur, and are handled as specified in [RFC3550] and
[RFC5576], resulting in the synchronisation source identifier of the
affecting RTP sources and/or sinks changing. An RTP source that
changes its RTP Session identifier (e.g. source transport address)
during a session has to choose a new SSRC identifier to avoid being
interpreted as looped source.
Synchronisation source identifiers that belong to the same
synchronisation context (i.e., that represent media streams that can
be synchronised using information in RTCP SR packets) are indicated
by use of identical CNAME chunks in corresponding RTCP SDES packets.
SDP signalling can also be used to provide explicit grouping of
synchronisation sources [RFC5576].
In some cases, the same SSRC Identifier value is used to relate
streams in two different RTP Sessions, such as in Multi-Session
Transmission of scalable video [RFC6190]. This is NOT RECOMMENDED
since there is no guarantee of uniqueness in SSRC values across
RTP sessions.
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Note that RTP sequence number and RTP timestamp are scoped by the
synchronisation source. Each RTP source will have a different
synchronisation source, and the corresponding media stream will have
a separate RTP sequence number and timestamp space.
An SSRC identifier is used by different type of sources as well as
sinks:
Real Media Source: Connected to a "physical" media source, for
example a camera or microphone.
Processed Media Source: A source with some attributed property
generated by some network node, for example a filtering function
in an RTP mixer that provides the most active speaker based on
some criteria, or a mix representing a set of other sources.
RTP Sink: A source that does not generate any RTP media stream in
itself (e.g. an endpoint or middlebox only receiving in an RTP
session). It still needs a sender SSRC for use as source in RTCP
reports.
Note that a endpoint that generates more than one media type, e.g. a
conference participant sending both audio and video, need not (and
commonly does not) use the same SSRC value across RTP sessions. RTCP
Compound packets containing the CNAME SDES item is the designated
method to bind an SSRC to a CNAME, effectively cross-correlating
SSRCs within and between RTP Sessions as coming from the same
endpoint. The main property attributed to SSRCs associated with the
same CNAME is that they are from a particular synchronisation context
and can be synchronised at playback.
An RTP receiver receiving a previously unseen SSRC value will
interpret it as a new source. It might in fact be a previously
existing source that had to change SSRC number due to an SSRC
conflict. However, the originator of the previous SSRC ought to have
ended the conflicting source by sending an RTCP BYE for it prior to
starting to send with the new SSRC, so the new SSRC is anyway
effectively a new source.
4.3. Contributing Source (CSRC)
The Contributing Source (CSRC) is not a separate identifier. Rather
a synchronisation source identifier is listed as a CSRC in the RTP
header of a packet generated by an RTP mixer if the corresponding
SSRC was in the header of one of the packets that contributed to the
mix.
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It is not possible, in general, to extract media represented by an
individual CSRC since it is typically the result of a media mixing
(merge) operation by an RTP mixer on the individual media streams
corresponding to the CSRC identifiers. The exception is the case
when only a single CSRC is indicated as this represent forwarding of
a media stream, possibly modified. The RTP header extension for
Mixer-to-Client Audio Level Indication [RFC6465] expands on the
receivers information about a packet with a CSRC list. Due to these
restrictions, CSRC will not be considered a fully qualified
multiplexing point and will be disregarded in the rest of this
document.
4.4. RTP Payload Type
Each Media Stream utilises one or more RTP payload formats. An RTP
payload format describes how the output of a particular media codec
is framed and encoded into RTP packets. The payload format used is
identified by the payload type field in the RTP data packet header.
The combination therefore identifies a specific Media Stream encoding
format. The format definition can be taken from [RFC3551] for
statically allocated payload types, but ought to be explicitly
defined in signalling, such as SDP, both for static and dynamic
Payload Types. The term "format" here includes whatever can be
described by out-of-band signalling means. In SDP, the term "format"
includes media type, RTP timestamp sampling rate, codec, codec
configuration, payload format configurations, and various robustness
mechanisms such as redundant encodings [RFC2198].
The payload type is scoped by sending endpoint within an RTP Session.
All synchronisation sources sent from an single endpoint share the
same payload types definitions. The RTP Payload Type is designed
such that only a single Payload Type is valid at any time instant in
the RTP source's RTP timestamp time line, effectively time-
multiplexing different Payload Types if any change occurs. The
payload type used can change on a per-packet basis for an SSRC, for
example a speech codec making use of generic comfort noise [RFC3389].
If there is a true need to send multiple Payload Types for the same
SSRC that are valid for the same instant, then redundant encodings
[RFC2198] can be used. Several additional constraints than the ones
mentioned above need to be met to enable this use, one of which is
that the combined payload sizes of the different Payload Types ought
not exceed the transport MTU.
Other aspects of RTP payload format use are described in RTP Payload
HowTo [I-D.ietf-payload-rtp-howto].
The payload type is not a multiplexing point at the RTP layer (see
Appendix A for a detailed discussion of why using the payload type as
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an RTP multiplexing point does not work). The RTP payload type is,
however, used to determine how to render a media stream, and so can
be viewed as selecting a rendering context. The rendering context
can be defined by the signalling, and the RTP payload type number is
sometimes used to associate an RTP media stream with the signalling.
This association is possible provided unique RTP payload type numbers
are used in each context. For example, an RTP media stream can be
associated with an SDP "m=" line by comparing the RTP payload type
numbers used by the media stream with payload types signalled in the
"a=rtpmap:" lines in the media sections of the SDP. If RTP media
streams are being associated with signalling contexts based on the
RTP payload type, then the assignment of RTP payload type numbers
MUST be unique across signalling contexts; if the same RTP payload
format configuration is used in multiple contexts, then a different
RTP payload type number has to be assigned in each context to ensure
uniqueness. If the RTP payload type number is not being used to
associated RTP media streams with a signalling context, then the same
RTP payload type number can be used to indicate the exact same RTP
payload format configuration in multiple contexts.
5. RTP Topologies and Issues
The impact of how RTP multiplexing is performed will in general vary
with how the RTP Session participants are interconnected, described
by RTP Topology [RFC5117] and its intended successor
[I-D.westerlund-avtcore-rtp-topologies-update].
5.1. Point to Point
Even the most basic use case, denoted Topo-Point-to-Point in
[I-D.westerlund-avtcore-rtp-topologies-update], raises a number of
considerations that are discussed in detail below (Section 6). They
range over such aspects as:
o Does my communication peer support RTP as defined with multiple
SSRCs?
o Do I need network differentiation in form of QoS?
o Can the application more easily process and handle the media
streams if they are in different RTP sessions?
o Do I need to use additional media streams for RTP retransmission
or FEC.
o etc.
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The point to point topology can contain one to many RTP sessions with
one to many media sources per session, each having one or more RTP
sources per media source.
5.2. Translators & Gateways
A point to point communication can end up in a situation when the
peer it is communicating with is not compatible with the other peer
for various reasons:
o No common media codec for a media type thus requiring transcoding
o Different support for multiple RTP sources and RTP sessions
o Usage of different media transport protocols, i.e RTP or other.
o Usage of different transport protocols, e.g. UDP, DCCP, TCP
o Different security solutions, e.g. IPsec, TLS, DTLS, SRTP with
different keying mechanisms.
In many situations this is resolved by the inclusion of a translator
between the two peers, as described by Topo-PtP-Translator in
[I-D.westerlund-avtcore-rtp-topologies-update]. The translator's
main purpose is to make the peer look to the other peer like
something it is compatible with. There can also be other reasons
than compatibility to insert a translator in the form of a middlebox
or gateway, for example a need to monitor the media streams. If the
stream transport characteristics are changed by the translator,
appropriate media handling can require thorough understanding of the
application logic, specifically any congestion control or media
adaptation.
5.3. Point to Multipoint Using Multicast
The Point to Multi-point topology is using Multicast to interconnect
the session participants. This includes both Topo-ASM and Topo-SSM
in [I-D.westerlund-avtcore-rtp-topologies-update].
Special considerations need to be made as multicast is a one to many
distribution system. For example, the only practical method for
adapting the bit-rate sent towards a given receiver for large groups
is to use a set of multicast groups, where each multicast group
represents a particular bit-rate. Otherwise the whole group gets
media adapted to the participant with the worst conditions. The
media encoding is either scalable, where multiple layers can be
combined, or simulcast, where a single version is selected. By
either selecting or combing multicast groups, the receiver can
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control the bit-rate sent on the path to itself. It is also common
that streams that improve transport robustness are sent in their own
multicast group to allow for interworking with legacy or to support
different levels of protection.
The result of this is some common behaviours for RTP multicast:
1. Multicast applications use a group of RTP sessions, not one.
Each endpoint will need to be a member of a number of RTP
sessions in order to perform well.
2. Within each RTP session, the number of RTP Sinks is likely to be
much larger than the number of RTP sources.
3. Multicast applications need signalling functions to identify the
relationships between RTP sessions.
4. Multicast applications need signalling functions to identify the
relationships between SSRCs in different RTP sessions.
All multicast configurations share a signalling requirement; all of
the participants will need to have the same RTP and payload type
configuration. Otherwise, A could for example be using payload type
97 as the video codec H.264 while B thinks it is MPEG-2. It is to be
noted that SDP offer/answer [RFC3264] is not appropriate for ensuring
this property. The signalling aspects of multicast are not explored
further in this memo.
Security solutions for this type of group communications are also
challenging. First of all the key-management and the security
protocol needs to support group communication. Source authentication
requires special solutions. For more discussion on this please
review Options for Securing RTP Sessions
[I-D.ietf-avtcore-rtp-security-options].
5.4. Point to Multipoint Using an RTP Transport Translator
This mode is described as Topo-Translator in
[I-D.westerlund-avtcore-rtp-topologies-update].
Transport Translators (Relays) result in an RTP session situation
that is very similar to how an ASM group RTP session would behave.
One of the most important aspects with the simple relay is that it is
only rewriting transport headers, no RTP modifications nor media
transcoding occur. The most obvious downside of this basic relaying
is that the translator has no control over how many streams need to
be delivered to a receiver. Nor can it simply select to deliver only
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certain streams, as this creates session inconsistencies: If the
translator temporarily stops a stream, this prevents some receivers
from reporting on it. From the sender's perspective it will look
like a transport failure. Applications needing to stop or switch
streams in the central node ought to consider using an RTP mixer to
avoid this issue.
The Transport Translator has the same signalling requirement as
multicast: All participants need to have the same payload type
configuration. Most of the ASM security issues also arise here.
Some alternatives when it comes to solution do exist, as there exists
a central node to communicate with, one that also can enforce some
security policies depending on the level of trust placed in the node.
5.5. Point to Multipoint Using an RTP Mixer
A mixer, described by Topo-Mixer in
[I-D.westerlund-avtcore-rtp-topologies-update], is a centralised node
that selects or mixes content in a conference to optimise the RTP
session so that each endpoint only needs connect to one entity, the
mixer. The media sent from the mixer to the endpoint can be
optimised in different ways. These optimisations include methods
like only choosing media from the currently most active speaker or
mixing together audio so that only one audio stream is needed.
Mixers have some downsides, the first is that the mixer has to be a
trusted node as they repacketize the media, and can perform media
transformation operations. When using SRTP, both media operations
and repacketization requires that the mixer verifies integrity,
decrypts the content, performs the operation and forms new RTP
packets, encrypts and integrity-protects them. This applies to all
types of mixers. The second downside is that all these operations
and optimisations of the session requires processing. How much
depends on the implementation, as will become evident below.
A mixer, unlike a pure transport translator, is always application
specific: the application logic for stream mixing or stream selection
has to be embedded within the mixer, and controlled using application
specific signalling. The implementation of a mixer can take several
different forms, as discussed below.
A Mixer can also contain translator functionalities, like a media
transcoder to adjust the media bit-rate or codec used for a
particular RTP media stream.
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Using multiple media streams is a well supported feature of RTP.
However, it can be unclear for most implementers or people writing
RTP/RTCP applications or extensions attempting to apply multiple
streams when it is most appropriate to add an additional SSRC in an
existing RTP session and when it is better to use multiple RTP
sessions. This section tries to discuss the various considerations
needed. The next section then concludes with some guidelines.
6.1. RTP and RTCP Protocol Considerations
This section discusses RTP and RTCP aspects worth considering when
selecting between using an additional SSRC and Multiple RTP sessions.
6.1.1. The RTP SpecificationRFC 3550 contains some recommendations and a bullet list with 5
arguments for different aspects of RTP multiplexing. Let's review
Section 5.2 of [RFC3550], reproduced below:
"For efficient protocol processing, the number of multiplexing points
should be minimised, as described in the integrated layer processing
design principle [ALF]. In RTP, multiplexing is provided by the
destination transport address (network address and port number) which
is different for each RTP session. For example, in a teleconference
composed of audio and video media encoded separately, each medium
SHOULD be carried in a separate RTP session with its own destination
transport address.
Separate audio and video streams SHOULD NOT be carried in a single
RTP session and demultiplexed based on the payload type or SSRC
fields. Interleaving packets with different RTP media types but
using the same SSRC would introduce several problems:
1. If, say, two audio streams shared the same RTP session and the
same SSRC value, and one were to change encodings and thus
acquire a different RTP payload type, there would be no general
way of identifying which stream had changed encodings.
2. An SSRC is defined to identify a single timing and sequence
number space. Interleaving multiple payload types would require
different timing spaces if the media clock rates differ and would
require different sequence number spaces to tell which payload
type suffered packet loss.
3. The RTCP sender and receiver reports (see Section 6.4) can only
describe one timing and sequence number space per SSRC and do not
carry a payload type field.
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4. An RTP mixer would not be able to combine interleaved streams of
incompatible media into one stream.
5. Carrying multiple media in one RTP session precludes: the use of
different network paths or network resource allocations if
appropriate; reception of a subset of the media if desired, for
example just audio if video would exceed the available bandwidth;
and receiver implementations that use separate processes for the
different media, whereas using separate RTP sessions permits
either single- or multiple-process implementations.
Using a different SSRC for each medium but sending them in the same
RTP session would avoid the first three problems but not the last
two.
On the other hand, multiplexing multiple related sources of the same
medium in one RTP session using different SSRC values is the norm for
multicast sessions. The problems listed above don't apply: an RTP
mixer can combine multiple audio sources, for example, and the same
treatment is applicable for all of them. It might also be
appropriate to multiplex streams of the same medium using different
SSRC values in other scenarios where the last two problems do not
apply."
Let's consider one argument at a time. The first is an argument for
using different SSRC for each individual media stream, which is very
applicable.
The second argument is advocating against using payload type
multiplexing, which still stands as can been seen by the extensive
list of issues found in Appendix A.
The third argument is yet another argument against payload type
multiplexing.
The fourth is an argument against multiplexing media streams that
require different handling into the same session. As we saw in the
discussion of RTP mixers, the RTP mixer has to embed application
logic in order to handle streams anyway; the separation of streams
according to stream type is just another piece of application logic,
which might or might not be appropriate for a particular application.
A type of application that can mix different media sources "blindly"
is the audio only "telephone" bridge; most other type of application
needs application-specific logic to perform the mix correctly.
The fifth argument discusses network aspects that we will discuss
more below in Section 6.3. It also goes into aspects of
implementation, like decomposed endpoints where different processes
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or inter-connected devices handle different aspects of the whole
multi-media session.
A summary of RFC 3550's view on multiplexing is to use unique SSRCs
for anything that is its own media/packet stream, and to use
different RTP sessions for media streams that don't share a media
type. This document supports the first point; it is very valid. The
later is one thing which is further discussed in this document as
something the application developer needs to make a conscious choice
for, but where imposing a single solution on all usages of RTP is
inappropriate.
6.1.1.1. Different Media Types: Recommendations
The above quote from RTP [RFC3550] includes a strong recommendation:
"For example, in a teleconference composed of audio and video
media encoded separately, each medium SHOULD be carried in a
separate RTP session with its own destination transport address."
It was identified in "Why RTP Sessions Should Be Content Neutral"
[I-D.alvestrand-rtp-sess-neutral] that the above statement is poorly
supported by any of the motivations provided in the RTP
specification. This has resulted in the creation of a specification
Multiple Media Types in an RTP Session specification
[I-D.ietf-avtcore-multi-media-rtp-session] which intends to update
this recommendation. That document has a detailed analysis of the
potential issues in having multiple media types in the same RTP
session. This document tries to provide an moreover arching
consideration regarding the usage of RTP session and considers
multiple media types in one RTP session as possible choice for the
RTP application designer.
6.1.2. Multiple SSRCs in a Session
Using multiple SSRCs in an RTP session at one endpoint requires
resolving some unclear aspects of the RTP specification. These could
potentially lead to some interoperability issues as well as some
potential significant inefficencies. These are further discussed in
"RTP Considerations for Endpoints Sending Multiple Media Streams"
[I-D.lennox-avtcore-rtp-multi-stream]. A application designer needs
to consider these issues and the impact availability or lack of the
optimization in the endpoints has on their application.
If an application will become affected by the issues described, using
Multiple RTP sessions can mitigate these issues.
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In some applications, the set of simultaneously active sources varies
within a larger set of session members. A receiver can then possibly
try to use a set of decoding chains that is smaller than the number
of senders, switching the decoding chains between different senders.
As each media decoding chain can contain state, either the receiver
needs to either be able to save the state of swapped-out senders, or
the sender needs to be able to send data that permits the receiver to
reinitialise when it resumes activity.
This behaviour will cause similar issues independent of Additional
SSRC or Multiple RTP session.
6.1.4. Cross Session RTCP Requests
There currently exists no functionality to make truly synchronised
and atomic RTCP messages with some type of request semantics across
multiple RTP Sessions. Instead, separate RTCP messages will have to
be sent in each session. This gives streams in the same RTP session
a slight advantage as RTCP messages for different streams in the same
session can be sent in a compound RTCP packet, thus providing an
atomic operation if different modifications of different streams are
requested at the same time.
When using multiple RTP sessions, the RTCP timing rules in the
sessions and the transport aspects, such as packet loss and jitter,
prevents a receiver from relying on atomic operations, forcing it to
use more robust and forgiving mechanisms.
6.1.5. Binding Related Sources
A common problem in a number of various RTP extensions has been how
to bind related RTP sources and their media streams together. This
issue is common to both using additional SSRCs and Multiple RTP
sessions.
The solutions can be divided into some groups, RTP/RTCP based,
Signalling based (SDP), grouping related RTP sessions, and grouping
SSRCs within an RTP session. Most solutions are explicit, but some
implicit methods have also been applied to the problem.
The SDP-based signalling solutions are:
SDP Media Description Grouping: The SDP Grouping Framework [RFC5888]
uses various semantics to group any number of media descriptions.
These has previously been considered primarily as grouping RTP
sessions, but this might change.
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SDP SSRC grouping: Source-Specific Media Attributes in SDP [RFC5576]
includes a solution for grouping SSRCs the same way as the
Grouping framework groupes Media Descriptions.
SDP MSID grouping: Media Stream Identifiers [I-D.ietf-mmusic-msid]
includes a solution for grouping SSRCs that is independent of
their allocation to RTP sessions.
This supports a lot of use cases. All these solutions have
shortcomings in cases where the session's dynamic properties are such
that it is difficult or resource consuming to keep the list of
related SSRCs up to date.
Within RTP/RTCP based solutions when binding to a endpoint or
synchronization context, i.e. the CNAME has not be sufficient and
one has multiple RTP sessions has been to using the same SSRC value
across all the RTP sessions. RTP Retransmission [RFC4588] is
multiple RTP session mode, Generic FEC [RFC5109], as well as the RTP
payload format for Scalable Video Coding [RFC6190] in Multi Session
Transmission (MST) mode uses this method. This method clearly works
but might have some downside in RTP sessions with many participating
SSRCs. The birthday paradox ensures that if you populate a single
session with 9292 SSRCs at random, the chances are approximately 1%
that at least one collision will occur. When a collision occur this
will force one to change SSRC in all RTP sessions and thus
resynchronizing all of them instead of only the single media stream
having the collision.
It can be noted that Section 8.3 of the RTP Specification [RFC3550]
recommends using a single SSRC space across all RTP sessions for
layered coding.
Another solution that has been applied to binding SSRCs has been an
implicit method used by RTP Retransmission [RFC4588] when doing
retransmissions in the same RTP session as the source RTP media
stream. This issues an RTP retransmission request, and then await a
new SSRC carrying the RTP retransmission payload and where that SSRC
is from the same CNAME. This limits a requestor to having only one
outstanding request on any new source SSRCs per endpoint.
There exists no RTP/RTCP based mechanism capable of supporting
explicit association accross multiple RTP sessions as well within an
RTP session. A proposed solution for handling this issue is
[I-D.westerlund-avtext-rtcp-sdes-srcname]. If accepted, this can
potentially also be part of an SDP based solution also by reusing the
same identifiers and name space.
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There exist a number of Forward Error Correction (FEC) based schemes
for how to reduce the packet loss of the original streams. Most of
the FEC schemes will protect a single source flow. The protection is
achieved by transmitting a certain amount of redundant information
that is encoded such that it can repair one or more packet losses
over the set of packets they protect. This sequence of redundant
information also needs to be transmitted as its own media stream, or
in some cases instead of the original media stream. Thus many of
these schemes create a need for binding related flows as discussed
above. Looking at the history of these schemes, there are schemes
using multiple SSRCs and schemes using multiple RTP sessions, and
some schemes that support both modes of operation.
Using multiple RTP sessions supports the case where some set of
receivers might not be able to utilise the FEC information. By
placing it in a separate RTP session, it can easily be ignored.
In usages involving multicast, having the FEC information on its own
multicast group, and therefore in its own RTP session, allows for
flexibility. This is especially useful when receivers see very
heterogeneous packet loss rates. Those receivers that are not seeing
packet loss don't need to join the multicast group with the FEC data,
and so avoid the overhead of receiving unnecessary FEC packets, for
example.
6.1.7. Transport Translator Sessions
A basic Transport Translator relays any incoming RTP and RTCP packets
to the other participants. The main difference between Additional
SSRCs and Multiple RTP Sessions resulting from this use case is that
with Additional SSRCs it is not possible for a particular session
participant to decide to receive a subset of media streams. When
using separate RTP sessions for the different sets of media streams,
a single participant can choose to leave one of the sessions but not
the other.
6.2. Interworking Considerations
There are several different kinds of interworking, and this section
discusses two related ones. The interworking between different
applications and the implications of potentially different choices of
usage of RTP's multiplexing points. The second topic relates to what
limitations have to be considered working with some legacy
applications.
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It is not uncommon that applications or services of similar usage,
especially the ones intended for interactive communication, encounter
a situation where one want to interconnect two or more of these
applications.
In these cases one ends up in a situation where one might use a
gateway to interconnect applications. This gateway then needs to
change the multiplexing structure or adhere to limitations in each
application.
There are two fundamental approaches to gatewaying: RTP Translator
interworking (RTP bridging), where the gateway acts as an RTP
Translator, and the two applications are members of the same RTP
session, and Gateway Interworking (with RTP termination), where there
are independent RTP sessions running from each interconnected
application to the gateway.
6.2.2. RTP Translator Interworking
From an RTP perspective the RTP Translator approach could work if all
the applications are using the same codecs with the same payload
types, have made the same multiplexing choices, have the same
capabilities in number of simultaneous media streams combined with
the same set of RTP/RTCP extensions being supported. Unfortunately
this might not always be true.
When one is gatewaying via an RTP Translator, a natural requirement
is that the two applications being interconnected need to use the
same approach to multiplexing. Furthermore, if one of the
applications is capable of working in several modes (such as being
able to use Additional SSRCs or Multiple RTP sessions at will), and
the other one is not, successful interconnection depends on locking
the more flexible application into the operating mode where
interconnection can be successful, even if no participants using the
less flexible application are present when the RTP sessions are being
created.
6.2.3. Gateway Interworking
When one terminates RTP sessions at the gateway, there are certain
tasks that the gateway has to carry out:
o Generating appropriate RTCP reports for all media streams
(possibly based on incoming RTCP reports), originating from SSRCs
controlled by the gateway.
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o Handling SSRC collision resolution in each application's RTP
sessions.
o Signalling, choosing and policing appropriate bit-rates for each
session.
If either of the applications has any security applied, e.g. in the
form of SRTP, the gateway needs to be able to decrypt incoming
packets and re-encrypt them in the other application's security
context. This is necessary even if all that's needed is a simple
remapping of SSRC numbers. If this is done, the gateway also needs
to be a member of the security contexts of both sides, of course.
Other tasks a gateway might need to apply include transcoding (for
incompatible codec types), rescaling (for incompatible video size
requirements), suppression of content that is known not to be handled
in the destination application, or the addition or removal of
redundancy coding or scalability layers to fit the need of the
destination domain.
From the above, we can see that the gateway needs to have an intimate
knowledge of the application requirements; a gateway is by its nature
application specific, not a commodity product.
This fact reveals the potential for these gateways to block evolution
of the applications by blocking unknown RTP and RTCP extensions that
the regular application has been extended with.
If one uses security functions, like SRTP, they can as seen above
incur both additional risk due to the gateway needing to be in
security association between the endpoints, unless the gateway is on
the transport level, and additional complexities in form of the
decrypt-encrypt cycles needed for each forwarded packet. SRTP, due
to its keying structure, also requires that each RTP session needs
different master keys, as use of the same key in two RTP sessions can
result in two-time pads that completely breaks the confidentiality of
the packets.
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Historically, the most common RTP use cases have been point to point
Voice over IP (VoIP) or streaming applications, commonly with no more
than one media source per endpoint and media type (typically audio
and video). Even in conferencing applications, especially voice
only, the conference focus or bridge has provided a single stream
with a mix of the other participants to each participant. It is also
common to have individual RTP sessions between each endpoint and the
RTP mixer, meaning that the mixer functions as an RTP-terminating
gateway.
When establishing RTP sessions that can contain endpoints that aren't
updated to handle multiple streams following these recommendations, a
particular application can have issues with multiple SSRCs within a
single session. These issues include:
1. Need to handle more than one stream simultaneously rather than
replacing an already existing stream with a new one.
2. Be capable of decoding multiple streams simultaneously.
3. Be capable of rendering multiple streams simultaneously.
This indicates that gateways attempting to interconnect to this class
of devices has to make sure that only one media stream of each type
gets delivered to the endpoint if it's expecting only one, and that
the multiplexing format is what the device expects. It is highly
unlikely that RTP translator-based interworking can be made to
function successfully in such a context.
6.3. Network Considerations
The multiplexing choice has impact on network level mechanisms that
need to be considered by the implementor.
6.3.1. Quality of Service
When it comes to Quality of Service mechanisms, they are either flow
based or marking based. RSVP [RFC2205] is an example of a flow based
mechanism, while Diff-Serv [RFC2474] is an example of a Marking based
one. For a marking based scheme, the method of multiplexing will not
affect the possibility to use QoS.
However, for a flow based scheme there is a clear difference between
the methods. Additional SSRC will result in all media streams being
part of the same 5-tuple (protocol, source address, destination
address, source port, destination port) which is the most common
selector for flow based QoS. Thus, separation of the level of QoS
between media streams is not possible. That is however possible when
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using multiple RTP sessions, where each media stream for which a
separate QoS handling is desired can be in a different RTP session
that can be sent over different 5-tuples.
6.3.2. NAT and Firewall Traversal
In today's network there exist a large number of middleboxes. The
ones that normally have most impact on RTP are Network Address
Translators (NAT) and Firewalls (FW).
Below we analyze and comment on the impact of requiring more
underlying transport flows in the presence of NATs and Firewalls:
End-Point Port Consumption: A given IP address only has 65536
available local ports per transport protocol for all consumers of
ports that exist on the machine. This is normally never an issue
for an end-user machine. It can become an issue for servers that
handle large number of simultaneous streams. However, if the
application uses ICE to authenticate STUN requests, a server can
serve multiple endpoints from the same local port, and use the
whole 5-tuple (source and destination address, source and
destination port, protocol) as identifier of flows after having
securely bound them to the remote endpoint address using the STUN
request. In theory the minimum number of media server ports
needed are the maximum number of simultaneous RTP Sessions a
single endpoint can use. In practice, implementation will
probably benefit from using more server ports to simplify
implementation or avoid performance bottlenecks.
NAT State: If an endpoint sits behind a NAT, each flow it generates
to an external address will result in a state that has to be kept
in the NAT. That state is a limited resource. In home or Small
Office/Home Office (SOHO) NATs, memory or processing are usually
the most limited resources. For large scale NATs serving many
internal endpoints, available external ports are likely the scarce
resource. Port limitations is primarily a problem for larger
centralised NATs where endpoint independent mapping requires each
flow to use one port for the external IP address. This affects
the maximum number of internal users per external IP address.
However, it is worth pointing out that a real-time video
conference session with audio and video is likely using less than
10 UDP flows, compared to certain web applications that can use
100+ TCP flows to various servers from a single browser instance.
NAT Traversal Excess Time: Making the NAT/FW traversal takes a
certain amount of time for each flow. It also takes time in a
phase of communication between accepting to communicate and the
media path being established which is fairly critical. The best
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case scenario for how much extra time it takes after finding the
first valid candidate pair following the specified ICE procedures
are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing
timer, which ICE specifies to be no smaller than 20 ms. That
assumes a message in one direction, and then an immediate
triggered check back. The reason it isn't more, is that ICE first
finds one candidate pair that works prior to attempting to
establish multiple flows. Thus, there is no extra time until one
has found a working candidate pair. Based on that working pair
the needed extra time is to in parallel establish the, in most
cases 2-3, additional flows. However, packet loss causes extra
delays, at least 100 ms, which is the minimal retransmission timer
for ICE.
NAT Traversal Failure Rate: Due to the need to establish more than a
single flow through the NAT, there is some risk that establishing
the first flow succeeds but that one or more of the additional
flows fail. The risk that this happens is hard to quantify, but
ought to be fairly low as one flow from the same interfaces has
just been successfully established. Thus only rare events such as
NAT resource overload, or selecting particular port numbers that
are filtered etc, ought to be reasons for failure.
Deep Packet Inspection and Multiple Streams: Firewalls differ in how
deeply they inspect packets. There exist some potential that
deeply inspecting firewalls will have similar legacy issues with
multiple SSRCs as some stack implementations.
Additional SSRC keeps the additional media streams within one RTP
Session and transport flow and does not introduce any additional NAT
traversal complexities per media stream. This can be compared with
normally one or two additional transport flows per RTP session when
using multiple RTP sessions. Additional lower layer transport flows
will be needed, unless an explicit de-multiplexing layer is added
between RTP and the transport protocol. A proposal for how to
multiplex multiple RTP sessions over the same single lower layer
transport exist in [I-D.westerlund-avtcore-transport-multiplexing].
6.3.3. Multicast
Multicast groups provides a powerful semantics for a number of real-
time applications, especially the ones that desire broadcast-like
behaviours with one endpoint transmitting to a large number of
receivers, like in IPTV. But that same semantics do result in a
certain number of limitations.
One limitation is that for any group, sender side adaptation to the
actual receiver properties causes degradation for all participants to
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what is supported by the receiver with the worst conditions among the
group participants. In most cases this is not acceptable. Instead
various receiver based solutions are employed to ensure that the
receivers achieve best possible performance. By using scalable
encoding and placing each scalability layer in a different multicast
group, the receiver can control the amount of traffic it receives.
To have each scalability layer on a different multicast group, one
RTP session per multicast group is used.
In addition, the transport flow considerations in multicast are a bit
different from unicast; NATs are not useful in the multicast
environment, meaning that the entire port range of each multicast
address is available for distinguishing between RTP sessions.
Thus it appears easiest and most straightforward to use multiple RTP
sessions for sending different media flows used for adapting to
network conditions.
6.3.4. Multiplexing multiple RTP Session on a Single Transport
For applications that don't need flow based QoS and like to save
ports and NAT/FW traversal costs and where usage of multiple media
types in one RTP session is not suitable, there is a proposal for how
to achieve multiplexing of multiple RTP sessions over the same lower
layer transport [I-D.westerlund-avtcore-transport-multiplexing].
Using such a solution would allow Multiple RTP session without most
of the perceived downsides of Multiple RTP sessions creating a need
for additional transport flows, but this solution would require
support from all functions that handle RTP packets, including
firewalls.
6.4. Security and Key Management Considerations
When dealing with point-to-point, 2-member RTP sessions only, there
are few security issues that are relevant to the choice of having one
RTP session or multiple RTP sessions. However, there are a few
aspects of multiparty sessions that might warrant consideration. For
general information of possible methods of securing RTP, please
review RTP Security Options [I-D.ietf-avtcore-rtp-security-options].
6.4.1. Security Context Scope
When using SRTP [RFC3711] the security context scope is important and
can be a necessary differentiation in some applications. As SRTP's
crypto suites (so far) are built around symmetric keys, the receiver
will need to have the same key as the sender. This results in that
no one in a multi-party session can be certain that a received packet
really was sent by the claimed sender or by another party having
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access to the key. In most cases this is a sufficient security
property, but there are a few cases where this does create issues.
The first case is when someone leaves a multi-party session and one
wants to ensure that the party that left can no longer access the
media streams. This requires that everyone re-keys without
disclosing the keys to the excluded party.
A second case is when using security as an enforcing mechanism for
differentiation. Take for example a scalable layer or a high quality
simulcast version which only premium users are allowed to access.
The mechanism preventing a receiver from getting the high quality
stream can be based on the stream being encrypted with a key that
user can't access without paying premium, having the key-management
limit access to the key.
SRTP [RFC3711] has no special functions for dealing with different
sets of master keys for different SSRCs. The key-management
functions have different capabilities to establish different set of
keys, normally on a per endpoint basis. For example, DTLS-SRTP
[RFC5764] and Security Descriptions [RFC4568] establish different
keys for outgoing and incoming traffic from an endpoint. This key
usage has to be written into the cryptographic context, possibly
associated with different SSRCs.
6.4.2. Key Management for Multi-party session
Performing key-management for multi-party session can be a challenge.
This section considers some of the issues.
Multi-party sessions, such as transport translator based sessions and
multicast sessions, cannot use Security Description [RFC4568] nor
DTLS-SRTP [RFC5764] without an extension as each endpoint provides
its set of keys. In centralised conferences, the signalling
counterpart is a conference server and the media plane unicast
counterpart (to which DTLS messages would be sent) is the transport
translator. Thus an extension like Encrypted Key Transport
[I-D.ietf-avt-srtp-ekt] is needed or a MIKEY [RFC3830] based solution
that allows for keying all session participants with the same master
key.
6.4.3. Complexity Implications
The usage of security functions can surface complexity implications
of the choice of multiplexing and topology. This becomes especially
evident in RTP topologies having any type of middlebox that processes
or modifies RTP/RTCP packets. Where there is very small overhead for
an RTP translator or mixer to rewrite an SSRC value in the RTP packet
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of an unencrypted session, the cost of doing it when using
cryptographic security functions is higher. For example if using
SRTP [RFC3711], the actual security context and exact crypto key are
determined by the SSRC field value. If one changes it, the
encryption and authentication tag needs to be performed using another
key. Thus changing the SSRC value implies a decryption using the old
SSRC and its security context followed by an encryption using the new
one.
7. Archetypes
This section discusses some archetypes of how RTP multiplexing can be
used in applications to achieve certain goals and a summary of their
implications. For each archetype there is discussion of benefits and
downsides.
7.1. Single SSRC per Session
In this archetype each endpoint in a point-to-point session has only
a single SSRC, thus the RTP session contains only two SSRCs, one
local and one remote. This session can be used both unidirectional,
i.e. only a single media stream or bi-directional, i.e. both
endpoints have one media stream each. If the application needs
additional media flows between the endpoints, they will have to
establish additional RTP sessions.
The Pros:
1. This archetype has great legacy interoperability potential as it
will not tax any RTP stack implementations.
2. The signalling has good possibilities to negotiate and describe
the exact formats and bit-rates for each media stream, especially
using today's tools in SDP.
3. It does not matter if usage or purpose of the media stream is
signalled on media stream level or session level as there is no
difference.
4. It is possible to control security association per RTP media
stream with current key-management, since each media stream is
directly related to an RTP session, and the keying operates on a
per-session basis.
The Cons:
a. The number of RTP sessions grows directly in proportion with the
number of media streams, which has the implications:
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* Linear growth of the amount of NAT/FW state with number of
media streams.
* Increased delay and resource consumption from NAT/FW
traversal.
* Likely larger signalling message and signalling processing
requirement due to the amount of session related information.
* Higher potential for a single media stream to fail during
transport between the endpoints.
b. When the number of RTP sessions grows, the amount of explicit
state for relating media stream also grows, linearly or possibly
exponentially, depending on how the application needs to relate
media streams.
c. The port consumption might become a problem for centralised
services, where the central node's port consumption grows rapidly
with the number of sessions.
d. For applications where the media streams are highly dynamic in
their usage, i.e. entering and leaving, the amount of signalling
can grow high. Issues arising from the timely establishment of
additional RTP sessions can also arise.
e. Cross session RTCP requests might be needed, and the fact that
they're impossible can cause issues.
f. If the same SSRC value is reused in multiple RTP sessions rather
than being randomly chosen, interworking with applications that
uses another multiplexing structure than this application will
require SSRC translation.
g. Cannot be used with Any Source Multicast (ASM) as one cannot
guarantee that only two endpoints participate as packet senders.
Using SSM, it is possible to restrict to these requirements if no
RTCP feedback is injected back into the SSM group.
h. For most security mechanisms, each RTP session or transport flow
requires individual key-management and security association
establishment thus increasing the overhead.
RTP applications that need to inter-work with legacy RTP
applications, like most deployed VoIP and video conferencing
solutions, can potentially benefit from this structure. However, a
large number of media descriptions in SDP can also run into issues
with existing implementations. For any application needing a larger
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number of media flows, the overhead can become very significant.
This structure is also not suitable for multi-party sessions, as any
given media stream from each participant, although having same usage
in the application, needs its own RTP session. In addition, the
dynamic behaviour that can arise in multi-party applications can tax
the signalling system and make timely media establishment more
difficult.
7.2. Multiple SSRCs of the Same Media Type
In this archetype, each RTP session serves only a single media type.
The RTP session can contain multiple media streams, either from a
single endpoint or from multiple endpoints. This commonly creates a
low number of RTP sessions, typically only one for audio and one for
video, with a corresponding need for two listening ports when using
RTP/RTCP multiplexing.
The Pros:
1. Low number of RTP sessions needed compared to single SSRC case.
This implies:
* Reduced NAT/FW state
* Lower NAT/FW Traversal Cost in both processing and delay.
2. Allows for early de-multiplexing in the processing chain in RTP
applications where all media streams of the same type have the
same usage in the application.
3. Works well with media type de-composite endpoints.
4. Enables Flow-based QoS with different prioritisation between
media types.
5. For applications with dynamic usage of media streams, i.e. they
come and go frequently, having much of the state associated with
the RTP session rather than an individual SSRC can avoid the need
for in-session signalling of meta-information about each SSRC.
6. Low overhead for security association establishment.
The Cons:
a. May have some need for cross session RTCP requests for things
that affect both media types in an asynchronous way.
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b. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per endpoint.
c. Will not be able to control security association for sets of
media streams within the same media type with today's key-
management mechanisms, unless these are split into different RTP
sessions.
For RTP applications where all media streams of the same media type
share same usage, this structure provides efficiency gains in amount
of network state used and provides more fate sharing with other media
flows of the same type. At the same time, it is still maintaining
almost all functionalities when it comes to negotiation in the
signalling of the properties for the individual media type and also
enabling flow based QoS prioritisation between media types. It
handles multi-party session well, independently of multicast or
centralised transport distribution, as additional sources can
dynamically enter and leave the session.
7.3. Multiple Sessions for one Media type
In this archetype one goes one step further than in the above
(Section 7.2) by using multiple RTP sessions also for a single media
type, but still not as far as having a single SSRC per RTP session.
The main reason for going in this direction is that the RTP
application needs separation of the media streams due to their usage.
Some typical reasons for going to this archetype are scalability over
multicast, simulcast, need for extended QoS prioritisation of media
streams due to their usage in the application, or the need for fine-
grained signalling using today's tools.
The Pros:
1. More suitable for Multicast usage where receivers can
individually select which RTP sessions they want to participate
in, assuming each RTP session has its own multicast group.
2. Indication of the application's usage of the media stream, where
multiple different usages exist.
3. Less need for SSRC specific explicit signalling for each media
stream and thus reduced need for explicit and timely signalling.
4. Enables detailed QoS prioritisation for flow based mechanisms.
5. Works well with de-composite endpoints.
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6. Handles dynamic usage of media streams well.
7. For transport translator based multi-party sessions, this
structure allows for improved control of which type of media
streams an endpoint receives.
8. The scope for who is included in a security association can be
structured around the different RTP sessions, thus enabling such
functionality with existing key-management.
The Cons:
a. Increases the amount of RTP sessions compared to Multiple SSRCs
of the Same Media Type.
b. Increased amount of session configuration state.
c. May need synchronised cross-session RTCP requests and require
some consideration due to this.
d. For media streams that are part of scalability, simulcast or
transport robustness it will be needed to bind sources, which
need to support multiple RTP sessions.
e. Some potential for concern with legacy implementations that does
not support the RTP specification fully when it comes to handling
multiple SSRC per endpoint.
f. Higher overhead for security association establishment.
g. If the applications need finer control than on media type level
over which session participants that are included in different
sets of security associations, most of today's key-management
will have difficulties establishing such a session.
For more complex RTP applications that have several different usages
for media streams of the same media type and / or uses scalability or
simulcast, this solution can enable those functions at the cost of
increased overhead associated with the additional sessions. This
type of structure is suitable for more advanced applications as well
as multicast based applications requiring differentiation to
different participants.
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Internet-Draft Guidelines for Multiplexing in RTP July 20137.4. Multiple Media Types in one Session
This archetype is to use a single RTP session for multiple different
media types, like audio and video, and possibly also transport
robustness mechanisms like FEC or Retransmission. Each media stream
will use its own SSRC and a given SSRC value from a particular
endpoint will never use the SSRC for more than a single media type.
The Pros:
1. Single RTP session which implies:
* Minimal NAT/FW state.
* Minimal NAT/FW Traversal Cost.
* Fate-sharing for all media flows.
2. Enables separation of the different media types based on the
payload types so media type specific endpoint or central
processing can still be supported despite single session.
3. Can handle dynamic allocations of media streams well on an RTP
level. Depends on the application's needs for explicit
indication of the stream usage and how timely that can be
signalled.
4. Minimal overhead for security association establishment.
The Cons:
a. Less suitable for interworking with other applications that uses
individual RTP sessions per media type or multiple sessions for a
single media type, due to need of SSRC translation.
b. Negotiation of bandwidth for the different media types is
currently not possible in SDP. This requires SDP extensions to
enable payload or source specific bandwidth. Likely to be a
problem due to media type asymmetry in needed bandwidth.
c. Not suitable for de-composite endpoints.
d. Flow based QoS cannot provide separate treatment to some media
streams compared to others in the single RTP session.
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e. If there is significant asymmetry between the media streams' RTCP
reporting needs, there are some challenges in configuration and
usage to avoid wasting RTCP reporting on the media stream that
does not need that frequent reporting.
f. Not suitable for applications where some receivers like to
receive only a subset of the media streams, especially if
multicast or transport translator is being used.
g. Additional concern with legacy implementations that do not
support the RTP specification fully when it comes to handling
multiple SSRC per endpoint, as also multiple simultaneous media
types needs to be handled.
h. If the applications need finer control over which session
participants that are included in different sets of security
associations, most key-management will have difficulties
establishing such a session.
7.5. Summary
There are some clear relations between these archetypes. Both the
"single SSRC per RTP session" and the "multiple media types in one
session" are cases which require full explicit signalling of the
media stream relations. However, they operate on two different
levels where the first primarily enables session level binding, and
the second needs to do it all on SSRC level. From another
perspective, the two solutions are the two extreme points when it
comes to number of RTP sessions needed.
The two other archetypes "Multiple SSRCs of the Same Media Type" and
"Multiple Sessions for one Media Type" are examples of two other
cases that first of all allows for some implicit mapping of the role
or usage of the media streams based on which RTP session they appear
in. It thus potentially allows for less signalling and in particular
reduced need for real-time signalling in dynamic sessions. They also
represent points in between the first two when it comes to amount of
RTP sessions established, i.e. representing an attempt to reduce the
amount of sessions as much as possible without compromising the
functionality the session provides both on network level and on
signalling level.
8. Summary considerations and guidelines8.1. Guidelines
This section contains a number of recommendations for implementors or
specification writers when it comes to handling multi-stream.
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Do not Require the same SSRC across Sessions: As discussed in
Section 6.1.5 there exist drawbacks in using the same SSRC in
multiple RTP sessions as a mechanism to bind related media streams
together. It is instead suggested that a mechanism to explicitly
signal the relation is used, either in RTP/RTCP or in the used
signalling mechanism that establishes the RTP session(s).
Use additional SSRCs additional Media Sources: In the cases where an
RTP endpoint needs to transmit additional media streams of the
same media type in the application, with the same processing
requirements at the network and RTP layers, it is suggested to
send them as additional SSRCs in the same RTP session. For
example a telepresence room where there are three cameras, and
each camera captures 2 persons sitting at the table, sending each
camera as its own SSRC within a single RTP session is suggested.
Use additional RTP sessions for streams with different requirements:
When media streams have different processing requirements from the
network or the RTP layer at the endpoints, it is suggested that
the different types of streams are put in different RTP sessions.
This includes the case where different participants want different
subsets of the set of RTP streams.
When using multiple RTP Sessions use grouping: When using Multiple
RTP session solutions, it is suggested to explicitly group the
involved RTP sessions when needed using the signalling mechanism,
for example The Session Description Protocol (SDP) Grouping
Framework. [RFC5888], using some appropriate grouping semantics.
RTP/RTCP Extensions May Support Additional SSRCs as well as Multiple RTP sessions:
When defining an RTP or RTCP extension, the creator needs to
consider if this extension is applicable to usage with additional
SSRCs and Multiple RTP sessions. Any extension intended to be
generic is suggested to support both. Applications that are not
as generally applicable will have to consider if interoperability
is better served by defining a single solution or providing both
options.
Transport Support Extensions: When defining new RTP/RTCP extensions
intended for transport support, like the retransmission or FEC
mechanisms, they are expected to include support for both
additional SSRCs and multiple RTP sessions so that application
developers can choose freely from the set of mechanisms without
concerning themselves with which of the multiplexing choices a
particular solution supports.
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[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6190] Wenger, S., Wang, Y.-K., Schierl, T., and A.
Eleftheriadis, "RTP Payload Format for Scalable Video
Coding", RFC 6190, May 2011.
[RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for
Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)", RFC 6222, April 2011.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax,
"Unicast-Based Rapid Acquisition of Multicast RTP
Sessions", RFC 6285, June 2011.
[RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time
Transport Protocol (RTP) Header Extension for Mixer-to-
Client Audio Level Indication", RFC 6465, December 2011.
Appendix A. Dismissing Payload Type Multiplexing
This section documents a number of reasons why using the payload type
as a multiplexing point for most things related to multiple streams
is unsuitable. If one attempts to use Payload type multiplexing
beyond it's defined usage, that has well known negative effects on
RTP. To use Payload type as the single discriminator for multiple
streams implies that all the different media streams are being sent
with the same SSRC, thus using the same timestamp and sequence number
space. This has many effects:
1. Putting restraint on RTP timestamp rate for the multiplexed
media. For example, media streams that use different RTP
timestamp rates cannot be combined, as the timestamp values need
to be consistent across all multiplexed media frames. Thus
streams are forced to use the same rate. When this is not
possible, Payload Type multiplexing cannot be used.
2. Many RTP payload formats can fragment a media object over
multiple packets, like parts of a video frame. These payload
formats need to determine the order of the fragments to
correctly decode them. Thus it is important to ensure that all
fragments related to a frame or a similar media object are
transmitted in sequence and without interruptions within the
object. This can relatively simple be solved on the sender side
by ensuring that the fragments of each media stream are sent in
sequence.
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3. Some media formats require uninterrupted sequence number space
between media parts. These are media formats where any missing
RTP sequence number will result in decoding failure or invoking
of a repair mechanism within a single media context. The text/
T140 payload format [RFC4103] is an example of such a format.
These formats will need a sequence numbering abstraction
function between RTP and the individual media stream before
being used with Payload Type multiplexing.
4. Sending multiple streams in the same sequence number space makes
it impossible to determine which Payload Type and thus which
stream a packet loss relates to.
5. If RTP Retransmission [RFC4588] is used and there is a loss, it
is possible to ask for the missing packet(s) by SSRC and
sequence number, not by Payload Type. If only some of the
Payload Type multiplexed streams are of interest, there is no
way of telling which missing packet(s) belong to the interesting
stream(s) and all lost packets need be requested, wasting
bandwidth.
6. The current RTCP feedback mechanisms are built around providing
feedback on media streams based on stream ID (SSRC), packet
(sequence numbers) and time interval (RTP Timestamps). There is
almost never a field to indicate which Payload Type is reported,
so sending feedback for a specific media stream is difficult
without extending existing RTCP reporting.
7. The current RTCP media control messages [RFC5104] specification
is oriented around controlling particular media flows, i.e.
requests are done addressing a particular SSRC. Such mechanisms
would need to be redefined to support Payload Type multiplexing.
8. The number of payload types are inherently limited.
Accordingly, using Payload Type multiplexing limits the number
of streams that can be multiplexed and does not scale. This
limitation is exacerbated if one uses solutions like RTP and
RTCP multiplexing [RFC5761] where a number of payload types are
blocked due to the overlap between RTP and RTCP.
9. At times, there is a need to group multiplexed streams and this
is currently possible for RTP Sessions and for SSRC, but there
is no defined way to group Payload Types.
10. It is currently not possible to signal bandwidth requirements
per media stream when using Payload Type Multiplexing.
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11. Most existing SDP media level attributes cannot be applied on a
per Payload Type level and would require re-definition in that
context.
12. A legacy endpoint that doesn't understand the indication that
different RTP payload types are different media streams might be
slightly confused by the large amount of possibly overlapping or
identically defined RTP Payload Types.
Appendix B. Proposals for Future Work
The above discussion and guidelines indicates that a small set of
extension mechanisms could greatly improve the situation when it
comes to using multiple streams independently of Multiple RTP session
or Additional SSRC. These extensions are:
Media Source Identification: A Media source identification that can
be used to bind together media streams that are related to the
same media source. A proposal
[I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES
item SRCNAME that also can be used with the a=ssrc SDP attribute
to provide signalling layer binding information.
MSID: A Media Stream identification scheme that can be used to
signal relationships between SSRCs that can be in the same or in
different RTP sessions. Described in [I-D.ietf-mmusic-msid]
SSRC limitations within RTP sessions: By providing a signalling
solution that allows the signalling peers to explicitly express
both support and limitations on how many simultaneous media
streams an endpoint can handle within a given RTP Session. That
ensures that usage of Additional SSRC occurs when supported and
without overloading an endpoint. This extension is proposed in
[I-D.westerlund-avtcore-max-ssrc].
Appendix C. Signalling considerations
Signalling is not an architectural consideration for RTP itself, so
this discussion has been moved to an appendix. However, it is hugely
important for anyone building complete applications, so it is
deserving of discussion.
The issues raised here need to be addressed in the WGs that deal with
signalling; they cannot be addressed by tweaking, extending or
profiling RTP.
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There exist various signalling solutions for establishing RTP
sessions. Many are SDP [RFC4566] based, however SDP functionality is
also dependent on the signalling protocols carrying the SDP. Where
RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative
fashion, while SIP [RFC3261] uses SDP with the additional definition
of Offer/Answer [RFC3264]. The impact on signalling and especially
SDP needs to be considered as it can greatly affect how to deploy a
certain multiplexing point choice.
C.1.1. Session Oriented Properties
One aspect of the existing signalling is that it is focused around
sessions, or at least in the case of SDP the media description.
There are a number of things that are signalled on a session level/
media description but those are not necessarily strictly bound to an
RTP session and could be of interest to signal specifically for a
particular media stream (SSRC) within the session. The following
properties have been identified as being potentially useful to signal
not only on RTP session level:
o Bitrate/Bandwidth exist today only at aggregate or a common any
media stream limit, unless either codec-specific bandwidth
limiting or RTCP signalling using TMMBR is used.
o Which SSRC that will use which RTP Payload Types (this will be
visible from the first media packet, but is sometimes useful to
know before packet arrival).
Some of these issues are clearly SDP's problem rather than RTP
limitations. However, if the aim is to deploy an solution using
additional SSRCs that contains several sets of media streams with
different properties (encoding/packetization parameter, bit-rate,
etc), putting each set in a different RTP session would directly
enable negotiation of the parameters for each set. If insisting on
additional SSRC only, a number of signalling extensions are needed to
clarify that there are multiple sets of media streams with different
properties and that they need in fact be kept different, since a
single set will not satisfy the application's requirements.
For some parameters, such as resolution and framerate, a SSRC-linked
mechanism has been proposed:
[I-D.lennox-mmusic-sdp-source-selection].
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Internet-Draft Guidelines for Multiplexing in RTP July 2013C.1.2. SDP Prevents Multiple Media Types
SDP chose to use the m= line both to delineate an RTP session and to
specify the top level of the MIME media type; audio, video, text,
image, application. This media type is used as the top-level media
type for identifying the actual payload format bound to a particular
payload type using the rtpmap attribute. This binding has to be
loosened in order to use SDP to describe RTP sessions containing
multiple MIME top level types.
There is an accepted WG item in the MMUSIC WG to define how multiple
media lines describe a single underlying transport
[I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible
in SDP to define one RTP session with media types having different
MIME top level types.
C.1.3. Signalling Media Stream Usage
Media streams being transported in RTP has some particular usage in
an RTP application. This usage of the media stream is in many
applications so far implicitly signalled. For example, an
application might choose to take all incoming audio RTP streams, mix
them and play them out. However, in more advanced applications that
use multiple media streams there will be more than a single usage or
purpose among the set of media streams being sent or received. RTP
applications will need to signal this usage somehow. The signalling
used will have to identify the media streams affected by their RTP-
level identifiers, which means that they have to be identified either
by their session or by their SSRC + session.
In some applications, the receiver cannot utilise the media stream at
all before it has received the signalling message describing the
media stream and its usage. In other applications, there exists a
default handling that is appropriate.
If all media streams in an RTP session are to be treated in the same
way, identifying the session is enough. If SSRCs in a session are to
be treated differently, signalling needs to identify both the session
and the SSRC.
If this signalling affects how any RTP central node, like an RTP
mixer or translator that selects, mixes or processes streams, treats
the streams, the node will also need to receive the same signalling
to know how to treat media streams with different usage in the right
fashion.
Authors' Addresses
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